webrtc/api
Mirko Bonadei 990d6b875e Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API"
This reverts commit 90bace0958.

Reason for revert: The original problem of this CL has been fixed in https://webrtc-review.googlesource.com/17540 but sounds like it is also adding voice_engine as a dependency of pc:peerconnection. We should investigate this because probably we can avoid it.

Original change's description:
> Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API
> 
> (this CL is based on the work by Taylor and Steve in https://webrtc-review.googlesource.com/c/src/+/10201)
> 
> This SetAudioPlayout method lets applications disable audio playout while
> still processing incoming audio data and generating statistics on the
> received audio.
> 
> This may be useful if the application wants to set up media flows as
> soon as possible, but isn't ready to play audio yet. Currently, native
> applications don't have any API point to control this, unless they
> completely implement their own AudioDeviceModule.
> 
> The SetAudioRecording works in a similar fashion but for the recorded
> audio. One difference is that calling SetAudioRecording(false) does not
> keep any audio processing alive.
> 
> TBR=solenberg
> 
> Bug: webrtc:7313
> Change-Id: I0aa075f6bfef9818f1080f85a8ff7842fb0750aa
> Reviewed-on: https://webrtc-review.googlesource.com/16180
> Reviewed-by: Henrik Andreassson <henrika@webrtc.org>
> Reviewed-by: Karl Wiberg <kwiberg@webrtc.org>
> Commit-Queue: Henrik Andreassson <henrika@webrtc.org>
> Cr-Commit-Position: refs/heads/master@{#20499}

TBR=solenberg@webrtc.org,henrika@webrtc.org,kwiberg@webrtc.org

Change-Id: I8431227e21dbffcfed3dd0e6bd7ce26c4ce09394
No-Presubmit: true
No-Tree-Checks: true
No-Try: true
Bug: webrtc:7313
Reviewed-on: https://webrtc-review.googlesource.com/17701
Reviewed-by: Mirko Bonadei <mbonadei@webrtc.org>
Commit-Queue: Mirko Bonadei <mbonadei@webrtc.org>
Cr-Commit-Position: refs/heads/master@{#20512}
2017-11-01 02:40:48 +00:00
..
audio Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
audio_codecs Hide the internal AudioEncoderOpus class by giving it an "Impl" suffix 2017-10-25 10:19:06 +00:00
call Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
ortc Adding NOLINT for typedefs.h and common_types.h 2017-09-15 13:03:51 +00:00
stats Add explicit includes of refcountedobject.h where it is used. 2017-10-06 13:00:14 +00:00
test Add StereoCodecAdapter classes 2017-10-31 06:39:52 +00:00
video Fix typo in VideoSendTiming header extension structure 2017-10-31 11:20:22 +00:00
video_codecs Reland "Add fine grained dropped video frames counters on sending side" 2017-10-25 09:32:15 +00:00
array_view.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
array_view_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
BUILD.gn Enable the clang style plugin in primary p2p/ target 2017-11-01 00:19:05 +00:00
candidate.cc Fix clang style warnings in api/candidate.h 2017-10-26 23:22:18 +00:00
candidate.h Fix clang style warnings in api/candidate.h 2017-10-26 23:22:18 +00:00
datachannel.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
datachannelinterface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
DEPS Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
dtmfsenderinterface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
fakemetricsobserver.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
fakemetricsobserver.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
jsep.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
jsepicecandidate.h Reland "Clean up libjingle API dependencies." 2017-10-05 13:51:21 +00:00
jsepsessiondescription.h Reland "Clean up libjingle API dependencies." 2017-10-05 13:51:21 +00:00
mediaconstraintsinterface.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mediaconstraintsinterface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mediastream.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mediastreaminterface.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mediastreaminterface.h Reland "Clean up libjingle API dependencies." 2017-10-05 13:51:21 +00:00
mediastreamproxy.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mediastreamtrack.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mediastreamtrackproxy.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mediatypes.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
mediatypes.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
notifier.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
optional.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
optional.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
optional_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
OWNERS Add phoglund@ to various OWNERS and remove kjellander@ 2017-10-19 09:21:12 +00:00
peerconnectionfactoryproxy.h Remove dead version of StartRtcEventLog 2017-10-06 15:18:24 +00:00
peerconnectioninterface.h Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" 2017-11-01 02:40:48 +00:00
peerconnectionproxy.h Revert "Add SetAudioPlayout and SetAudioRecording methods to the PeerConnection API" 2017-11-01 02:40:48 +00:00
proxy.cc Enable the clang style plugin in primary p2p/ target 2017-11-01 00:19:05 +00:00
proxy.h Enable the clang style plugin in primary p2p/ target 2017-11-01 00:19:05 +00:00
refcountedbase.h New classes RefCounter and RefCountedBase. 2017-10-23 11:46:47 +00:00
rtcerror.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtcerror.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtcerror_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtceventlogoutput.h Move RtcEventLogOutput to api/ 2017-10-06 13:58:14 +00:00
rtpparameters.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtpparameters.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtpparameters_unittest.cc Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtpreceiverinterface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtpsender.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
rtpsenderinterface.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
statstypes.cc Add explicit includes of refcountedobject.h where it is used. 2017-10-06 13:00:14 +00:00
statstypes.h Added RTCMediaStreamTrackStats.jitterBufferDelay for audio 2017-10-02 10:47:00 +00:00
streamcollection.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
turncustomizer.h TurnCustomizer - an interface for modifying stun messages sent by TurnPort 2017-10-11 07:45:29 +00:00
umametrics.cc Enable the clang style plugin in primary p2p/ target 2017-11-01 00:19:05 +00:00
umametrics.h Enable the clang style plugin in primary p2p/ target 2017-11-01 00:19:05 +00:00
videosourceproxy.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
videotracksource.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00
webrtcsdp.h Fixing WebRTC after moving from src/webrtc to src/ 2017-09-15 05:02:56 +00:00